In recent years, Voice over Internet Protocol (VoIP) has become increasingly popular for telephony communications. With this increasing popularity has come greater usage of the capacity of data networks for VoIP calls. Further, VoIP calls require relatively large amounts of bandwidth in a data network. It is therefore not surprising that data networks used for VoIP calls are more and more frequently operated at, and sometimes beyond, full capacity. Unfortunately, ways to measure the flow of data traffic through a network have not kept pace with the increased usage of VoIP communications. As discussed below, there is a need for improved systems and methods for measuring the bandwidth consumed by VoIP network traffic.
Indeed, assessments of call flows, including predictions relating to bandwidth consumption, are an important part of the network planning process that occurs before a network ever becomes operational. However, such assessments must be validated periodically, preferably on an ongoing basis, once a network becomes operational. In particular the load on each link in a network, i.e., the amount of bandwidth being consumed on each link compared to the amount of bandwidth available on the link, would be of great interest to a network operations center (NOC). Similarly, of great interest to the NOC would be the total amount of traffic being sent between two endpoints. Accordingly, being able to determine the bandwidth consumed by endpoint to endpoint call flows, and the loads these call flows place on the network, would be a valuable aid for successful network operation. Further, determining the bandwidth consumed by call flows would be useful for exploiting the deployed bandwidth capacity put into service in the network. As merely one example, such information would allow a network operator to more efficiently manage the network, including determining what parts of the network are being under-utilized and what parts of the network are in danger of being used beyond capacity.
As is well known, multiple network layers are used to transport data such as VoIP data. Those skilled in the art will understand that higher level network layers necessarily obtain information regarding network status, including utilization, from lower level network layers. Unfortunately, the present ability of Internet Protocol (IP) networks to communicate network status to the voice switching network layer and its call control plane is not well developed. As a result, NOCs must closely monitor the live IP network status that supports a network of softswitches in order to maintain service as intended. Presently, NOCs obtain information regarding network status in two ways. First, present mechanisms for monitoring the network may involve gathering network link and element status by means of Simple Network Management Protocol (SNMP) traps and gets. Such information may be used to populate alarm screens or network maps on a real-time or near real-time basis. Second, real-time or near real-time reports may be available for the softswitch as well. These reports may provide a running count of active calls (and/or signaled calls not yet active) intended for a VoIP network exclusively between softswitches and their media gateways, e.g., gateways between softswitches and other networks such as the Public Switched Telephone Network (PSTN).
Unfortunately, neither of the two ways of reporting on a network's status discussed above integrate and reconcile live IP network status with the live presented IP call loads from the voice switch media gateways that are connected at the edge of the IP network. Further, these presently available reports do not provide an understanding of call flows within the context of the current IP network configuration, nor do they present call flows in terms of the utilization bandwidth they consume aggregated according to the IP Wide Area Network (WAN) links involved. That is, presently available systems and methods do not examine the consequences to actual network IP routing conditions arising from routing such flows between their endpoints in the network. In sum, packet networks such as IP networks tend to be seen as “clouds” precisely because data flows, and the links that are used to transport data, are often too difficult to determine.
Accordingly, there is a need for improved ways for determining loads in a network carrying VoIP data traffic, both from the perspective of the load on individual links, as well as from the perspective of the overall load between two endpoints, e.g., two softswitches separated by two or more links. In particular, it would be desirable to provide a real time, unified visual view of a VoIP network that includes the status of network elements, e.g., information regarding whether each node was available, whether each link was available, and the utilization of each link. It would further be desirable, when viewing information regarding the utilization of a link, to be able to obtain real-time or near real time reporting on the traffic using the link to a first endpoint from a second endpoint. Such reporting capabilities would advantageously provide the ability to generate real-time or near real-time alarms upon the failure or potential over-utilization of a link. Such reporting capabilities would further advantageously provide the ability to generate recommendations for controlling and routing voice data received at endpoints based on failures and/or high utilizations in the IP network.